Voice over Internet Protocol (VoIP) represents a variety of different transmission technologies that are used to provide voice communications over Internet Protocol (IP) networks, such as the Internet or similar packet-switched networks. In a general VoIP-based example, a microphone converts sound into analog electrical signals. The analog signals are then converted to a digital form. If desired, compression techniques (e.g., audio codecs that encode speech) are used to reduce bandwidth requirements. The resulting data is formatted into Internet protocol (IP) packets for transmission over the Internet. The process is reversed at the receiving end eventually producing sound from a speaker.
The connection and disconnection processes that are implemented between two VoIP endpoint devices are sometimes referred to as set-up and tear-down, respectively. This set-up and tear-down of calls is implemented according to rules defined by various VoIP (and or video-based) session control protocols, such as, H.323, Session Initiation Protocol (SIP), Real-time Transport Protocol (RTP), and/or IP Multimedia Subsystem (IMS).
The growth of Voice-over-IP (VoIP) devices, services and products presents a number of issues. Call routing is an important consideration in terms of costs for VoIP capable devices. Other issues include call quality factors such as packet loss, packet delay, and packet jitter.
Another expanding area relates to Video-over-IP (VioIP) devices, services and products. Although the data requirements are at least partially different relative to VoIP, similar issues with quality and user experience arise with video transmissions. For instance, packet loss can cause intermittent freezing of a video stream as well as artifacts. Such aspects can be considerably frustrating to a viewer and even render the video stream unintelligible.